similar to: re. rtp.c RTP codec 19

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw

Hello. I am working on implementing RFC 5574 (RTP Payload Format for the Speex Codec) in the ffmpeg and have a question concerning it. It would be nice if somebody could answered it. * Chapter 4.1.1 Registration of Media Type Audio/Speex, subpart "Optional parameters" states these SDP optional parameters: vbr: variable bit-rate - either 'on', 'off', or 'vad'

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

All: Attached please find yet another RTP profile revision (v5). You can also find the document at: http://www.herlein.com/downloads/speex/docs/ Changes: - added vbr, cng, ebw, sr optional parameters to MIME - added vbr, cng, ebw a=fmtp options for SDP use - added required document attributes for submission to IETF and IANA (format and author contact information). Note that we

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

regarding my previous post about SIP outgoing calls dropping with an error 481 .. this is my output from a SIP debug. the call dropped occurs at the end. Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my control. help :) please!! Dave Signal=0 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

The latest revision of the draft profile for use of Speex in RTP is attached. We plan on submitting this - or a modified version of this, based on immediate feedback - to the IETF on Monday for consideration at the next meeting. Major differences in this revision are: - removed the discussion in the MIME section. It's a duplicate of the SDP discussion anyway, and may or may not match the

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing